Remove AudioClassifier

webrtc/modules/audio_coding/neteq/audio_classifier_unittest.cc

Index: webrtc/modules/audio_coding/neteq/audio_classifier_unittest.cc
diff --git a/webrtc/modules/audio_coding/neteq/audio_classifier_unittest.cc b/webrtc/modules/audio_coding/neteq/audio_classifier_unittest.cc
deleted file mode 100644
index 09c569966b7f8789e7659df185e4aa561bdf1980..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/neteq/audio_classifier_unittest.cc
+++ /dev/null
@@ -1,82 +0,0 @@
-/*
- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_coding/neteq/audio_classifier.h"
-
-#include <math.h>
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <memory>
-#include <string>
-
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
-
-namespace webrtc {
-
-static const size_t kFrameSize = 960;
-
-TEST(AudioClassifierTest, AllZeroInput) {
-  int16_t in_mono[kFrameSize] = {0};
-
-  // Test all-zero vectors and let the classifier converge from its default
-  // to the expected value.
-  AudioClassifier zero_classifier;
-  for (int i = 0; i < 100; ++i) {
-    zero_classifier.Analysis(in_mono, kFrameSize, 1);
-  }
-  EXPECT_TRUE(zero_classifier.is_music());
-}
-
-void RunAnalysisTest(const std::string& audio_filename,
-                     const std::string& data_filename,
-                     size_t channels) {
-  AudioClassifier classifier;
-  std::unique_ptr<int16_t[]> in(new int16_t[channels * kFrameSize]);
-  bool is_music_ref;
-
-  FILE* audio_file = fopen(audio_filename.c_str(), "rb");
-  ASSERT_TRUE(audio_file != NULL) << "Failed to open file " << audio_filename
-                                  << std::endl;
-  FILE* data_file = fopen(data_filename.c_str(), "rb");
-  ASSERT_TRUE(audio_file != NULL) << "Failed to open file " << audio_filename
-                                  << std::endl;
-  while (fread(in.get(), sizeof(int16_t), channels * kFrameSize, audio_file) ==
-         channels * kFrameSize) {
-    bool is_music =
-        classifier.Analysis(in.get(), channels * kFrameSize, channels);
-    EXPECT_EQ(is_music, classifier.is_music());
-    ASSERT_EQ(1u, fread(&is_music_ref, sizeof(is_music_ref), 1, data_file));
-    EXPECT_EQ(is_music_ref, is_music);
-  }
-  fclose(audio_file);
-  fclose(data_file);
-}
-
-TEST(AudioClassifierTest, DoAnalysisMono) {
-#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64)
-  RunAnalysisTest(test::ResourcePath("short_mixed_mono_48", "pcm"),
-                  test::ResourcePath("short_mixed_mono_48_arm", "dat"),
-                  1);
-#else
-  RunAnalysisTest(test::ResourcePath("short_mixed_mono_48", "pcm"),
-                  test::ResourcePath("short_mixed_mono_48", "dat"),
-                  1);
-#endif // WEBRTC_ARCH_ARM
-}
-
-TEST(AudioClassifierTest, DoAnalysisStereo) {
-  RunAnalysisTest(test::ResourcePath("short_mixed_stereo_48", "pcm"),
-                  test::ResourcePath("short_mixed_stereo_48", "dat"),
-                  2);
-}
-
-}  // namespace webrtc